Implements the GetSignalLevel and GetStats interface for the local audio track.

content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
index d8fae057c296a048d466fb1c96f6a52cd1a5234d..1cd7ad27d374a17773841d0cdcdd459dd89705b6 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h
@@ -20,6 +20,7 @@ class AudioRenderer;
 
 namespace webrtc {
 class AudioSourceInterface;
+class AudioProcessorInterface;
 }
 
 namespace content {
@@ -50,6 +51,13 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
   // level of the audio data.
   void SetSignalLevel(int signal_level);
 
+  // Method called by the WebRtcLocalAudioTrack to set the processor that
+  // applies signal processing on the data of the track.
+  // This class will keep a reference of the |processor|.
+  // Called on the main render thread.
+  void SetAudioProcessor(
+      talk_base::scoped_refptr<webrtc::AudioProcessorInterface> processor);

[tommi (sloooow) - chröme, 2014/03/04 17:33:54]

do we need to use talk_base::scoped_refptr here?
I'd prefer to use chrome types wherever possible and reduce usage of third_party
types to only where we absolutely need it.

[no longer working on chromium, 2014/03/04 18:54:07]

Done.

Actually it is not a problem since adapter is designed to work with the
libjingle interface.

+
  private:
   // webrtc::MediaStreamTrack implementation.
   virtual std::string kind() const OVERRIDE;
@@ -57,6 +65,9 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
   // webrtc::AudioTrackInterface implementation.
   virtual void AddSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
   virtual void RemoveSink(webrtc::AudioTrackSinkInterface* sink) OVERRIDE;
+  virtual bool GetSignalLevel(int* level) OVERRIDE;
+  virtual talk_base::scoped_refptr<webrtc::AudioProcessorInterface>
+      GetAudioProcessor() OVERRIDE;
 
   // cricket::AudioCapturer implementation.
   virtual void AddChannel(int channel_id) OVERRIDE;
@@ -73,6 +84,10 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
   // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
   talk_base::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
 
+  // The audio processsor that applies audio processing on the data of audio
+  // track.
+  talk_base::scoped_refptr<webrtc::AudioProcessorInterface> audio_processor_;

[tommi (sloooow) - chröme, 2014/03/04 17:33:54]

scoped_refptr here as well

[no longer working on chromium, 2014/03/04 18:54:07]

Done.

+
   // A vector of WebRtc VoE channels that the capturer sends data to.
   std::vector<int> voe_channels_;
 
@@ -80,7 +95,10 @@ class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
   // from the audio track.
   ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
 
-  // Protects |voe_channels_|.
+  // The amplitude of the signal.
+  int signal_level_;
+
+  // Protects |voe_channels_|, |audio_processor_| and |signal_level_|.
   mutable base::Lock lock_;
 };